Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(2869)

Unified Diff: webrtc/call/rtc_event_log_unittest.cc

Issue 1457023002: Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fix bug in RtcpPacketReportBlockTest Created 5 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/call/rtc_event_log_unittest.cc
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc
index 9dc7bec215ad26a33a5ec623bc2c4a72cbb92509..6e97a6634ae01f77c24eac0b9aca1813071e209c 100644
--- a/webrtc/call/rtc_event_log_unittest.cc
+++ b/webrtc/call/rtc_event_log_unittest.cc
@@ -16,6 +16,7 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/random.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread.h"
#include "webrtc/call.h"
@@ -23,7 +24,6 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/test/random.h"
#include "webrtc/test/test_suite.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
@@ -300,7 +300,7 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
uint32_t csrcs_count,
uint8_t* packet,
size_t packet_size,
- test::Random* prng) {
+ Random* prng) {
RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
Clock* clock = Clock::GetRealTimeClock();
@@ -348,7 +348,7 @@ size_t GenerateRtpPacket(uint32_t extensions_bitvector,
return header_size;
}
-rtc::scoped_ptr<rtcp::RawPacket> GenerateRtcpPacket(test::Random* prng) {
+rtc::scoped_ptr<rtcp::RawPacket> GenerateRtcpPacket(Random* prng) {
rtcp::ReportBlock report_block;
report_block.To(prng->Rand<uint32_t>()); // Remote SSRC.
report_block.WithFractionLost(prng->Rand(50));
@@ -365,7 +365,7 @@ rtc::scoped_ptr<rtcp::RawPacket> GenerateRtcpPacket(test::Random* prng) {
void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
VideoReceiveStream::Config* config,
- test::Random* prng) {
+ Random* prng) {
// Create a map from a payload type to an encoder name.
VideoReceiveStream::Decoder decoder;
decoder.payload_type = prng->Rand(0, 127);
@@ -394,7 +394,7 @@ void GenerateVideoReceiveConfig(uint32_t extensions_bitvector,
void GenerateVideoSendConfig(uint32_t extensions_bitvector,
VideoSendStream::Config* config,
- test::Random* prng) {
+ Random* prng) {
// Create a map from a payload type to an encoder name.
config->encoder_settings.payload_type = prng->Rand(0, 127);
config->encoder_settings.payload_name = (prng->Rand<bool>() ? "VP8" : "H264");
@@ -433,7 +433,7 @@ void LogSessionAndReadBack(size_t rtp_count,
VideoReceiveStream::Config receiver_config(nullptr);
VideoSendStream::Config sender_config(nullptr);
- test::Random prng(random_seed);
+ Random prng(random_seed);
// Create rtp_count RTP packets containing random data.
for (size_t i = 0; i < rtp_count; i++) {
@@ -590,7 +590,7 @@ TEST(RtcEventLogTest, LogSessionAndReadBack) {
1 + csrcs_count, // Number of BWE loss events.
extensions, // Bit vector choosing extensions.
csrcs_count, // Number of contributing sources.
- extensions + csrcs_count); // Random seed.
+ extensions * 3 + csrcs_count + 1); // Random seed.
}
}
}
@@ -608,7 +608,7 @@ void DropOldEvents(uint32_t extensions_bitvector,
VideoReceiveStream::Config receiver_config(nullptr);
VideoSendStream::Config sender_config(nullptr);
- test::Random prng(random_seed);
+ Random prng(random_seed);
// Create two RTP packets containing random data.
size_t packet_size = prng.Rand(1000, 1100);

Powered by Google App Engine
This is Rietveld 408576698